1. Field of the Invention
The present invention relates generally to digital filters, and more specifically, to a method and circuit for direct-form digital filtering that has improved overload recovery characteristics.
2. Background of the Invention
Direct-form digital filters are used in many applications including audio circuits such as equalizers and other processing blocks. In particular, direct form II (DF-II) filters provide efficient implementations that require only N delay stages for an Nth order filter, but may overload at internal summing nodes of the filter.
At some predetermined input operating level, and at certain frequencies, many digital filters will overload at some point in the filter that exceeds the ability of the filter elements to properly represent internal state variables and/or external (output) signal levels. In particular, fixed-point filters have a range that is limited by the number of bits used to express the values within the filter. The overload condition by definition causes internal distortion in a digital filter's poles, even when the clipping occurs at the output combiner, as feedback from the output combiner is used in implementing the poles.
In a digital filter, clipping causes a prolonged error period to occur that takes much longer for recovery than in a corresponding analog filter. The clipping changes the transfer function of the filter, moving corner or center frequencies away from their design values, whereas in analog filters, clipping may only cause distortion and some loss Q factor. The movement of the filter frequencies not only causes improper operation, but contributes to the filters' slow recovery or possible total lack of recovery from the overload condition.
Therefore, it would be desirable to provide a digital filtering circuit and method for digital filtering that recover from an overload condition quickly and with minimal disruption.